From 38fcd58429b29c9474f1647efed7ebeb543c0637 Mon Sep 17 00:00:00 2001 From: Jakob Ivarsson Date: Thu, 27 Oct 2022 22:38:57 +0200 Subject: [PATCH] Change default NetEq sample rate to 48k. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k). Bug: none Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662 Commit-Queue: Jakob Ivarsson‎ Reviewed-by: Henrik Lundin Cr-Commit-Position: refs/heads/main@{#38536} --- api/neteq/neteq.h | 2 +- modules/audio_coding/acm2/audio_coding_module_unittest.cc | 8 ++++---- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/api/neteq/neteq.h b/api/neteq/neteq.h index ffc3958345..5300c5601e 100644 --- a/api/neteq/neteq.h +++ b/api/neteq/neteq.h @@ -128,7 +128,7 @@ class NetEq { std::string ToString() const; - int sample_rate_hz = 16000; // Initial value. Will change with input data. + int sample_rate_hz = 48000; // Initial value. Will change with input data. bool enable_post_decode_vad = false; size_t max_packets_in_buffer = 200; int max_delay_ms = 0; diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc index 6d3ebbfd8d..7e4b764aed 100644 --- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc +++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc @@ -1030,7 +1030,7 @@ class AcmSenderBitExactnessNewApi : public AcmSenderBitExactnessOldApi {}; defined(NDEBUG) && defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480)); - Run(/*audio_checksum_ref=*/"a3077ac01b0137e8bbc237fb1f9816a5", + Run(/*audio_checksum_ref=*/"37ecdabad1698a857cf811e6d1fa91df", /*payload_checksum_ref=*/"3c79f16f34218271f3dca4e2b1dfe1bb", /*expected_packets=*/33, /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); @@ -1038,7 +1038,7 @@ TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) { TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960)); - Run(/*audio_checksum_ref=*/"76da9b7514f986fc2bb32b1c3170e8d4", + Run(/*audio_checksum_ref=*/"0e9078d23454901496a88362ba0740c3", /*payload_checksum_ref=*/"9e0a0ab743ad987b55b8e14802769c56", /*expected_packets=*/16, /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); @@ -1067,7 +1067,7 @@ TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) { TEST_F(AcmSenderBitExactnessOldApi, Pcm16_16000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160)); - Run(/*audio_checksum_ref=*/"bc6ab94d12a464921763d7544fdbd07e", + Run(/*audio_checksum_ref=*/"f95c87bdd33f631bcf80f4b19445bbd2", /*payload_checksum_ref=*/"ad786526383178b08d80d6eee06e9bad", /*expected_packets=*/100, /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); @@ -1151,7 +1151,7 @@ TEST_F(AcmSenderBitExactnessOldApi, Ilbc_30ms) { #if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) TEST_F(AcmSenderBitExactnessOldApi, G722_20ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("G722", 16000, 1, 9, 320, 160)); - Run(/*audio_checksum_ref=*/"a87a91ec0124510a64967f5d768554ff", + Run(/*audio_checksum_ref=*/"f5264affff25cf2cbd2e1e8a5217f9a3", /*payload_checksum_ref=*/"fc68a87e1380614e658087cb35d5ca10", /*expected_packets=*/50, /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput);